Gizmo along with asterisk cant dial a toll no or int'll no

I can receive calls through gizmo perfectly, can dial a toll free no, but when i dial a toll no the call does not complete





exten => 888,1,Answer()
exten => 888,n,Background(queue-thankyou)
exten => 888,n,Dial(SIP/1000)
exten => 888,n,Hangup()

exten => s,1,Answer()
exten => s,n,Goto(main_ivr,888,1)
exten => s,n,Hangup


exten => _1NXXXXXXXXX,1,SetCallerID("My Name" <1-My NO>)
exten => _1NXXXXXXXXX,n,Dial(SIP/${EXTEN},20,r)

exten => _011.,1,SetCallerID("My Name" <1-My NO>)
exten => _011.,n,Dial(SIP/${EXTEN:3},20,r)


exten => _1XXX,1,SayDigits(${EXTEN})
exten => _1XXX,n,Dial(SIP/${EXTEN},30)
exten => _1XXX,n,Playback(vm-nobodyavail)
exten => _1XXX,n,Hangup()

include => internal
include => outgoing
include => main_ivr
include => default

when I dial a no 1-425-635-3106 it gets failed with error 503 on my grandstream Phone

on the console it throws the following

SetCallerID(“SIP/1000-09de90b8”, ““My Name” <1-My NO>”) in new stack
– Executing [14256353106@phones:2] Dial(“SIP/1000-09de90b8”, “SIP/|20|r”) in new stack
– Called
[Aug 27 02:48:27] NOTICE[2258]: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to ‘"“My Name” sip:1MyNO@;tag=as0978a78b’
– SIP/ is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
== Auto fallthrough, channel ‘SIP/1000-09de90b8’ status is ‘CONGESTION’

when i dial international (011919810012345) call just drops

[Aug 27 02:52:00]
– Executing [011919810012345@phones:1] SetCallerID(“SIP/1000-09e179f8”, ““My Name” <1-My NO>”) in new stack
– Executing [011919810012345@phones:2] Dial(“SIP/1000-09e179f8”, “SIP/|20|r”) in new stack
– Called
– SIP/ answered SIP/1000-09e179f8
– Native bridging SIP/1000-09e179f8 and SIP/
== Spawn extension (phones, 011919810012345, 2) exited non-zero on ‘SIP/1000-09e179f8’

Please Help

Can Someone help